Log - 00d566ec297d52c8e732d7299d69e4e580707d61 - fp2-dev/platform/external/chromium_org/third_party/webrtc

0291c80 Removed ViE file API. by mflodman@webrtc.org · 11 years ago
20cfda6 Remove unused multi stream bandwidth estimator. by solenberg@webrtc.org · 11 years ago
b4c89a4 Making no NACK mode work again in VideoEngine. by mflodman@webrtc.org · 11 years ago
695ff2a Add support for padding in pacer. by stefan@webrtc.org · 11 years ago
54b6ebc Correctly set SSRCs for extra send RTP modules. by stefan@webrtc.org · 11 years ago
4e5f983 Fix a return value mismatch introduced in r4129. by stefan@webrtc.org · 11 years ago
6696fba Breaking out RTP header parsing from the RTP module. by stefan@webrtc.org · 11 years ago
f40e9b6 - Created RemoteBitrateEstimator wrapper for use internally in (ViE) ChannelGroup. by solenberg@webrtc.org · 11 years ago
eef4fd5 Adds integration test for RTX and fixes bugs found. by stefan@webrtc.org · 11 years ago
453f9c0 Add functions to ViE API to enable/disable the absolute send time header extension. by solenberg@webrtc.org · 11 years ago
281cff8 Include files from webrtc/.. paths in video_engine/ by pbos@webrtc.org · 11 years ago
ac6d919 Wiring down config from video engine until video coding and remote bitrate estimator modules instantiation. by andresp@webrtc.org · 11 years ago
06ad384 Trigger a PLI if the duration of non-decodable frames exceeds a threshold. by stefan@webrtc.org · 11 years ago
7bc7e02 Adding a payload type for RTX. by mflodman@webrtc.org · 11 years ago
67879bc WebRtc_Word32 -> int32_t in video_engine/ by pbos@webrtc.org · 11 years ago
208a648 Always set render delay in ViEChannel::RegisterExternalDecoder. by pbos@webrtc.org · 11 years ago
065b64d Remove UDP transport API from ViE by pwestin@webrtc.org · 11 years ago
946d240 Adding RTX on source by mikhal@webrtc.org · 11 years ago
912b7f7 Revert r3667 and r3665 Review URL: https://webrtc-codereview.appspot.com/1199004 by pwestin@webrtc.org · 11 years ago
b793abe Increasing size of nack list in buffered mode. by mikhal@webrtc.org · 11 years ago
2daec4c Removed the engine API:s related to transport such as SetSendDestination, the functionality is now provided via the test frame work. by pwestin@webrtc.org · 11 years ago
78e450f Enabling bufffering mode with no sync module or VoE by mikhal@webrtc.org · 11 years ago
55e6f58 Stop and restart fix. by mflodman@webrtc.org · 12 years ago
0329e59 Rename webrtc::StatsObserver to webrtc::CallStatsObserver by fischman@webrtc.org · 12 years ago
08998cd fixing nack list size calculation by mikhal@webrtc.org · 12 years ago
9d6fcb3 Adding a receive side API for buffering mode. by mikhal@webrtc.org · 12 years ago
0c66de6 Updates to send side streaming mode: by mikhal@webrtc.org · 12 years ago
cd1ac8b Don't report an error for GetEstimatedReceiveBandwidth if there is no valid by mflodman@webrtc.org · 12 years ago
1619664 Adding a send side API for streaming by mikhal@webrtc.org · 12 years ago
7fff32c Fix mismatch between different NACK list lengths and packet buffers. by stefan@webrtc.org · 12 years ago
2a5dbce Break out RemoteBitrateEstimator from RtpRtcp module and make RemoteBitrateEstimator::Process trigger new REMB messages. by stefan@webrtc.org · 12 years ago
f314c80 Added API to get receive side video delay. by mflodman@webrtc.org · 12 years ago
78696d3 Wire up CallStats to provide modules with correct RTT. by mflodman@webrtc.org · 12 years ago
5e87b5f Enable paced sender. Review URL: https://webrtc-codereview.appspot.com/965016 by pwestin@webrtc.org · 12 years ago
07e96da Reorganize modules/video_render. by andrew@webrtc.org · 12 years ago
b6d9cfc Revert the revert in r2988 since that wasn't the issue. by mflodman@webrtc.org · 12 years ago
9f269d2 Reverse Merged r2884 & r2888 from trunk. by vikasmarwaha@webrtc.org · 12 years ago
3bbed74 Switching to I420VideoFrame by mikhal@webrtc.org · 12 years ago
b015cbe Move src/ -> webrtc/ by andrew@webrtc.org · 12 years ago